similar to: PRI Call Redirection / Transfers

I think the solution may exist. The compatibility of mysql-connector-odbc with maria may just means the driver can access the mariadb - but my experience suggests not live on the same host. maria has its own connector: https://downloads.mariadb.org/connector-odbc/ it does not look like this is in the sig, so I'll have to turn to the maria repo. I'll try replacing the driver first, then

Thanks for each of your inputs. It was not a configuration issue as odbcinst.ini does not reference the mysql subdirectory. Rather than use Alexander's url, I ran: yum -y install centos-release-openstack-ocata yum -y install mariadb-server ...the cloud repository provides the properly pathed file. # repoquery -l mariadb-server mariadb-libs | grep lib64 | grep libmysqlclient

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Hi, I just got a T1 interface for a AT&T (became Lucent) System75 (uses same cards as Definity). I would take a crack at making a cable but can't determine the pinout for a cable and it is not apparent from the board. Asking you guys makes sense as one of you may have one of these systems. The cable has a amphenol male D50 connector on one end and probably a rj45 on the other. I also

I have a machine that crashes every so often. I believe the following macro is responsible (gotoif,$[${ARG3}] in particular). The macro works as expected: if ARG3 is defined - hop over assignment. But my hunch is that it gradually chews up memory. ; This macro is puts voicemail in an alternate mailbox (if ARG3 defined - otherwise Mailbox matches extension). [macro-stdexten] exten =>

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD

Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me. Thanks in advance. James Jones Signate, LLC james.jones@signate.com 415.442.4012 (office) 413.771.1402 (office) 413.977.6482 (mobile) 413.667.3105 (fax) 665 Third Street Suite

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN}) This says the digits the caller enters on the keypad, not the extension they are calling from. Thanks Guys!!!!!!!! Paul Paul Mahler pmahler@signate.com

If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/

Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

Sorry for the stupid question: What's the purpose of defining a peer as trunk in iax.conf ? The question is also valid generally speaking (for other channel types), for instance: why define a Zap group as trunk in extension.conf ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even

Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim